The background of the invention is discussed briefly in the following. In wireless IP multimedia networks (IP, Internet Protocol), such as the 3GPP Core Network IP Multimedia (3GPP, Third Generation Partnership Project), care must be taken to make efficient use of scarce resources such as the air interface.
As such, most of the codecs used in wireless terminals are narrowband audio and video codecs, such as the AMR-codec (AMR, Adaptive Multi-Rate), which is the default codec in 3G wireless networks (3G, Third Generation). The problem with such a narrowband codecs is that they do not allow DTMF tones to pass through (DTMF, Dual-Tone Multi-Frequency).
DTMF tones are widely used in interactive response systems, e.g. when the user is presented with a menu and has to make a selection. For instance, the user listens to an audio announcement that says “if you need assistance in English language press 1, if you need assistance in Spanish language press 2”.
In circuit switched networks, the selection is sent with DTMF tones. However, narrowband codecs, such as the AMR codec, will not be able to transport DTMF tones due to bandwidth constrains.
FIG. 1 illustrates a prior art arrangement for signalling in a wireless IP multimedia network. The presented prior art arrangement for signalling in a wireless IP multimedia network has a terminal A initiating a call to terminal B, marked with reference numbers 1 and 4 respectively.
The terminal A 1 initiating the call has a home network A 2 and while traveling connects to a visiting network A 3. For example, the home network A 2 can be in Stockholm, Sweden and the visiting network A 3 can be in Melbourne, Australia.
The terminal B 4 receiving the call has a home network B 5 and while traveling connects to a visiting network B 6. For example, the home network B 5 can be in Helsinki, Finland and the visiting network B 6 can be in Sydney, Australia.
The terminal A 1 in Melbourne initiates the call towards the terminal B 4 in Sydney. While initiating the call the signalling proceeds from the visiting network A 3 in Melbourne first to the home network A 2 in Stockholm, next to the home network B 5 in Helsinki and then to the visiting network B 6 in Sydney where the terminal B 4 is located.
When the signalling of the call initiation has been completed the speech connection itself can formed directly between the visiting network A 3 in Melbourne where the terminal A 1 is located and the visiting network B 6 in Sydney where the terminal B 4 is located.
Typically, when a multimedia session is established, a PDP context (PDP, Packet Data Protocol) for signalling e.g. SIP signalling is set-up (SIP, Session Initiation Protocol). The PDP context may be seen as a logical channel between the terminal and the GGSN (GGSN, Gateway GPRS Service Node), (GPRS, General Packet Radio System). In addition, a PDP context per media stream is set-up. Each PDP context includes certain quality of service and a certain protection against errors.
Those PDP contexts used for signalling are typically strongly protected against errors. Those PDP contexts used for user plane e.g. audio or video are typically not so strongly protected.
When the DTMF tones are sent in wireless IP multimedia networks there is a problem that the DTMF audio tone does not pass through a narrowband codec such as AMR-codec. Therefore, a solution based on sending a representation of the pressed digit, instead of an audio tone, is preferred.
It should be desirable to re-use an already existing PDP context (e.g., the one used for audio) and multiplex the DTMF digits with the actual audio data. However, when sending DTMF tones in wireless IP multimedia networks, it is a problem that as the PDP context used for audio may (typically will) be suffering weak and unequal error protection, this may cause to errors when a digit is transmitted over that PDP context. Small errors are acceptable for an audio channel, but not when the data is DTMF (e.g., a user press 1 and the media gateway receives 2 due to an error).
Furthermore, when sending DTMF tones in wireless IP multimedia networks, the delays in the signalling path cause additional problems. Another option is to re-use the PDP context allocated for signalling. This PDP context has a strong protection, so no errors are expected. However, this PDP context may be restricted to send and receive data to a particular entity in the network, e.g. a special SIP server known as the Proxy-CSCF (CSCF, Call State Control Function). However, the DTMF data will need to be received in an end-point, such as the Media Gateway. Usually, the IP address of the Media Gateway is not known in advance, and therefore, the PDP context used for signalling cannot be configured to allow transmission of data to a Media Gateway.
Another problem present in sending DTMF tones in wireless IP multimedia networks is that only a small portion of the calls uses DTMF signalling and as most of the calls don't use DTMF signalling they should not be affected. Therefore, setting-up a separate PDP context for all the calls, and for all the duration of the call seems to be not a reasonable solution, as in most cases the PDP context allocated to DTMF tones will not be, in general, used. Resources in the radio network must be efficiently used, and for the minimum duration time.
Furthermore, in wireless IP multimedia networks, the DTMF tones must not be sent over the call control signalling channel, e.g. SIP. The reason for this is that when inter-working with the PSTN (PSTN, Public Switched Telephone Network) occurs, the signalling channel e.g. SIP will terminate to a MGC (MGC, Media Gateway Controller), whereas the user plane will terminate to a MG (MG, Media Gateway). Tones are sent and received by the Media Gateway.
Furthermore, in wireless IP multimedia networks, the DTMF tones must not be sent over the call control signalling channel, e.g. SIP. Another reasons for not sending the DTMF tones over the call control signalling channel, e.g. SIP, is that the signalling channel will traverse a set of nodes in the home network. The user plane will follow the shortest path. In case of long-distance roaming, the signalling path will traverse a set of nodes separated by long distance, whereas the user plane may go to a next host. The synchronization between announcements in the audio channel (e.g., “press 1 for assistance in English”) and the actual keystrokes is hard to fulfil.